"use strict";
// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// Async-await-promise based SRS RTC Player.

function SrsRtcPlayerAsync() {
  var self = {};

  // @see https://github.com/rtcdn/rtcdn-draft
  // @url The WebRTC url to play with, for example:
  //      webrtc://r.ossrs.net/live/livestream
  // or specifies the API port:
  //      webrtc://r.ossrs.net:11985/live/livestream
  // or autostart the play:
  //      webrtc://r.ossrs.net/live/livestream?autostart=true
  // or change the app from live to myapp:
  //      webrtc://r.ossrs.net:11985/myapp/livestream
  // or change the stream from livestream to mystream:
  //      webrtc://r.ossrs.net:11985/live/mystream
  // or set the api server to myapi.domain.com:
  //      webrtc://myapi.domain.com/live/livestream
  // or set the candidate(eip) of answer:
  //      webrtc://r.ossrs.net/live/livestream?candidate=39.107.238.185
  // or force to access https API:
  //      webrtc://r.ossrs.net/live/livestream?schema=https
  // or use plaintext, without SRTP:
  //      webrtc://r.ossrs.net/live/livestream?encrypt=false
  // or any other information, will pass-by in the query:
  //      webrtc://r.ossrs.net/live/livestream?vhost=xxx
  //      webrtc://r.ossrs.net/live/livestream?token=xxx
  self.init = async function (url) {
    var conf = self.__internal.prepareUrl(url);
    self.pc.addTransceiver("audio", { direction: "recvonly" });
    self.pc.addTransceiver("video", { direction: "recvonly" });

    var offer = await self.pc.createOffer();
    await self.pc.setLocalDescription(offer);
    var session = await new Promise(function (resolve, reject) {
      // @see https://github.com/rtcdn/rtcdn-draft
      var data = {
        api: conf.apiUrl,
        tid: conf.tid,
        streamurl: conf.streamUrl,
        clientip: null,
        sdp: offer.sdp
      };
      // console.log("Generated offer: ", data);

      // $.ajax({
      // 	type: "POST",
      // 	url: conf.apiUrl,
      // 	data: JSON.stringify(data),
      // 	contentType: "application/json",
      // 	dataType: "json"
      // })
      // 	.done(function(data) {
      // 		console.log("Got answer: ", data);
      // 		if (data.code) {
      // 			reject(data);
      // 			return;
      // 		}

      // 		resolve(data);
      // 	})
      // 	.fail(function(reason) {
      // 		reject(reason);
      // 	});

      // 将JQ的ajax请求方式改成下面这个原生的
      // IE7+, Firefox, Chrome, Opera, Safari 或 IE6, IE5
      const xmlhttp = window.XMLHttpRequest ? new XMLHttpRequest() : new ActiveXObject("Microsoft.XMLHTTP");
      xmlhttp.onreadystatechange = function () {
        if (xmlhttp.readyState == 4 && xmlhttp.status == 200) {
          const data = JSON.parse(xmlhttp.response);
          if (data.code) {
            reject(data);
            return;
          }
          resolve(data);
        }
      };
      // console.log(conf.apiUrl)
      xmlhttp.open("POST", conf.apiUrl, true);
      xmlhttp.setRequestHeader(
        "Content-type",
        "application/x-www-form-urlencoded"
      );
      xmlhttp.send(JSON.stringify(data));

    });
    await self.pc.setRemoteDescription(
      new RTCSessionDescription({ type: "answer", sdp: session.sdp })
    );
    session.simulator =
      conf.schema +
      "//" +
      conf.urlObject.server +
      ":" +
      conf.port +
      "/rtc/v1/nack/";
    return session;
  };

  // Close the player.
  self.close = function () {
    self.pc && self.pc.close();
    self.pc = null;
  };

  // The callback when got remote track.
  // Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
  self.ontrack = function (event) {
    // https://webrtc.org/getting-started/remote-streams
    self.stream.addTrack(event.track);
  };

  // Internal APIs.
  self.__internal = {
    defaultPath: "/rtc/v1/play/",
    prepareUrl: function (webrtcUrl) {
      var urlObject = self.__internal.parse(webrtcUrl);

      // If user specifies the schema, use it as API schema.
      var schema = urlObject.user_query.schema;
      schema = schema ? schema + ":" : window.location.protocol;

      var port = urlObject.port || 1985;
      if (schema === "https:") {
        port = urlObject.port || 443;
      }

      // @see https://github.com/rtcdn/rtcdn-draft
      var api = urlObject.user_query.play || self.__internal.defaultPath;
      if (api.lastIndexOf("/") !== api.length - 1) {
        api += "/";
      }

      apiUrl = schema + "//" + urlObject.server + ":" + port + api;
      for (var key in urlObject.user_query) {
        if (key !== "api" && key !== "play") {
          apiUrl += "&" + key + "=" + urlObject.user_query[key];
        }
      }
      // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
      var apiUrl = apiUrl.replace(api + "&", api + "?");

      var streamUrl = urlObject.url;

      return {
        apiUrl: apiUrl,
        streamUrl: streamUrl,
        schema: schema,
        urlObject: urlObject,
        port: port,
        tid: Number(parseInt(new Date().getTime() * Math.random() * 100))
          .toString(16)
          .substr(0, 7)
      };
    },
    parse: function (url) {
      // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
      var a = document.createElement("a");
      a.href = url
        .replace("rtmp://", "http://")
        .replace("webrtc://", "http://")
        .replace("rtc://", "http://");

      var vhost = a.hostname;
      var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1);
      var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1);

      // parse the vhost in the params of app, that srs supports.
      app = app.replace("...vhost...", "?vhost=");
      if (app.indexOf("?") >= 0) {
        var params = app.substr(app.indexOf("?"));
        app = app.substr(0, app.indexOf("?"));

        if (params.indexOf("vhost=") > 0) {
          vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
          if (vhost.indexOf("&") > 0) {
            vhost = vhost.substr(0, vhost.indexOf("&"));
          }
        }
      }

      // when vhost equals to server, and server is ip,
      // the vhost is __defaultVhost__
      if (a.hostname === vhost) {
        var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
        if (re.test(a.hostname)) {
          vhost = "__defaultVhost__";
        }
      }

      // parse the schema
      var schema = "rtmp";
      if (url.indexOf("://") > 0) {
        schema = url.substr(0, url.indexOf("://"));
      }

      var port = a.port;
      if (!port) {
        if (schema === "http") {
          port = 80;
        } else if (schema === "https") {
          port = 443;
        } else if (schema === "rtmp") {
          port = 1935;
        }
      }

      var ret = {
        url: url,
        schema: schema,
        server: a.hostname,
        port: port,
        vhost: vhost,
        app: app,
        stream: stream
      };
      self.__internal.fill_query(a.search, ret);

      // For webrtc API, we use 443 if page is https, or schema specified it.
      if (!ret.port) {
        if (schema === "webrtc" || schema === "rtc") {
          if (ret.user_query.schema === "https") {
            ret.port = 443;
          } else if (window.location.href.indexOf("https://") === 0) {
            ret.port = 443;
          } else {
            // For WebRTC, SRS use 1985 as default API port.
            ret.port = 1985;
          }
        }
      }

      return ret;
    },
    fill_query: function (query_string, obj) {
      // pure user query object.
      obj.user_query = {};

      if (query_string.length === 0) {
        return;
      }

      // split again for angularjs.
      if (query_string.indexOf("?") >= 0) {
        query_string = query_string.split("?")[1];
      }

      var queries = query_string.split("&");
      for (var i = 0; i < queries.length; i++) {
        var elem = queries[i];

        var query = elem.split("=");
        obj[query[0]] = query[1];
        obj.user_query[query[0]] = query[1];
      }

      // alias domain for vhost.
      if (obj.domain) {
        obj.vhost = obj.domain;
      }
    }
  };

  self.pc = new RTCPeerConnection(null);

  // Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams
  self.stream = new MediaStream();

  // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
  self.pc.ontrack = function (event) {
    if (self.ontrack) {
      self.ontrack(event);
    }
  };

  return self;
}

window.SrsRtcPlayerAsync = SrsRtcPlayerAsync